similar to: Asterisk codec issue with sip / iax.

Dears; I do not know if any had experience in using speex or ilbc with IAX and got good results, because I am facing a problem with GSM. I am facing a noise problem when I am using GSM with IAX trunk as following: IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk using GSM codec ---> Remote Asterisk Box ---> Digium Card (FXO) to terminate the call to the destination While no

using asterisk(A) over iax to another asterisk server(B) which connects to pstn over pri. Doesn't B have translate to ulaw whatever goes out to the pstn, so therefore shouldn't A choose ulaw as the iax codec to B? That way there's no loss translating from {gsm, ildc, etc} to ulaw on the B server. My partner thinks I'm nuts, and that gsm is much more "efficient" as

Hi, I have strange problem. I have 1 SIP client (bt100) and 1 Iax2 client (IAXPhone): - when I call from Iax to SIP sound works - when I call from Sip to Iax sound doesn't work, I get : Jan 25 13:52:22 NOTICE[31334]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/200/1 of format gsm since our native format has changed to ulaw Why is Asterisk not satisfied with gsm

Hi all, I have spent some time searching, but I haven't found a way to prevent * from concatenating two frames into one IAX packet. I have a situation where I make an IAX GSM call to *, which transcodes to an iLBC SIP call. Every second voice packet the IAX client receives contains 2x 20ms frames, the other containing only one. I presume this is related to the mismatch of 20ms GSM vs

I've been told by someone at Digium that they discontinued the IAXy ATA. I was buying the X100P IAX ATA for some time until even they discontinued that. I am looking for a new ATA that does IAX2 and supports the following codecs: GSM, Speex, G711 Can anyone point me in the right direction? I'd prefer to stick with IAX protocol because it is easier to use working with NAT and SIP/NAT is

I have 2 Asterisk servers connected with IAX. It's working fine I can call an extension from one phone in an office to another phone in the other office. The only problem I have is lagging. What codec should I use? I have Cisco phones 7960 on both end. Currently in the IAX trunk I configured it to disallow all and use GSM only. In my sip config of each phone I use disallow all and allow

I seem to be having a problem that I have narrowed down to a disagreement on codec negotiation or codec setup of some kind in an IAX peering arrangement. Here's a non-ASCII art version of the setup: DID origination provider via SIP/gsm to Call routing asterisk server via IAX/gsm to Client asterisk server via SIP/ulaw to Polycom 501 UA The problem that occurs

Hello, I'd like to implement some public sip uri's that poeple can call into and get an echo test. Is there a way to force a codec so that users can test various codecs? Something like: echo-test at example.com (negotiates whatever codec, is there a way to figure out what codec was negotiated and tell the user) echo-test-g711 at example.com (forces g711) echo-test-g729 at

Hi, I have a problem with IAX accounts... I set up a few months ago an Asterisk server, with mysql support to load iax accounts. Settings seems fine because apparently the system works as expected. Yesterday I tried to add another iax account in the iax.conf directly. And I have a problem with this new account (named 444). I can authenticate from 444 to the server, and I can receive calls from

Thank you for your advise, I'll find something with a lot of memory.... Adrien -- Adrien Laurent - CIO 514-284-2020 adrien@modulis.ca www.modulis.ca -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jason Walker Sent: Saturday, October 01, 2005 12:25 AM To: 'Asterisk Users Mailing List - Non-Commercial

> On Thu, May 10, 2018 at 11:44:14AM -0700, Steve Edwards wrote: >> I receive an INVITE/SDP containing: >> >> m=audio 11310 RTP/AVP 3 0 101 >> >> which I interpret as gsm, ulaw, rfc2833. >> >> and I reply with an OK/SDP containing: >> >> m=audio 15884 RTP/AVP 0 3 101 >> >> which I interpret as ulaw, gsm, rfc2833. >>

hi We take calls inbound via SIP from our Cisco PSTN gateways, and pass it to customers using IAX (they run their own asterisk servers). We've noticed that asterisk is transcoding the call into a different codec, if the customer prefers a codec different to that which our cisco gw prefers. As such, the quality of the call can degrade. We'd rather asterisk just passed through the RTP

Hi everybody i have a problem trying to connect an incomming phone call from pstn to my (soft phone) iaxcomm, the phone rings but when i try to answer the call, asterisk sends a message like this. Jun 1 19:33:17 NOTICE[5013528]: channel.c:1223 ast_read: Dropping incompatible voice frame on IAX2[192.168.222.99:4569]/16 of format GSM since our native format has changed to ALAW i'm working

Hi, at my Asterisk box, I have a few of IAX2 phones (configured with alaw/ulaw/gsm codecs, in this order) and a PRI E1 line. In iax.conf I hav: disallow=all allow=alaw allow=ulaw allow=gsm During some incoming call, I read at console: -- Executing Dial("Zap/2-1", "IAX2/215|20|TtwW") in new stack -- Called 215 -- Call accepted by 10.97.1.7 (format ulaw) --

Hi list, I'm new with asterisk PBX and try to do what is described below : A B -------- Asterisk | PhoneA1 --- Internet -------- Phone B1 |--------------------- | PhoneA2 --- -------- Phone B2 Phones from both lans register in Asterisk. I 'd like that phones on both

Hi; I've just had a couple of discoveries in the learning process that are really making me impressed with this software. 1) IAX transfer - I'm running Asterisk boxes A, B, and C. B is in the middle and has a dialplan that points to extensions on C. When a client on A in the proper context on B tries dialling a client of C, B is smart enough to release itself somehow from playing

Hi, I have IAX trunks between Asterisk servers. They receive calls on ISDN cards and Dial() through the IAX trunks to the "primary" Asterisk server where all the SIP phone extensions are registered. The IAX trunk settings are something like this (all servers have this identical except for the "host" field): [inbound] deny=all allow=alaw allow=gsm type=friend

Hi All, I am wondering that how I can setup different codec for different extensions in my dial plan. scanario will when user X (Sip) call 111 extension in default context. The Asterisk response should be in GSM codec When user X (Sip) call 222 extension in default context. the Asterisk response should be in G711 Codec Actually I want to setup an extension for FAX receiving (rx_fax) and

Did anyone have a way to make codec negotiation work with Asterisk? This is something I would love to have working as well. I won't need PSTN -> G729 mixing. Just SIP -> SIP using G729 for calling remote offices via VPN, but everything else use G711. -----Original Message----- From: Brancaleoni Matteo [mailto:mbrancaleoni@espia.it] Sent: Wednesday, July 16, 2003 11:32 AM To:

HI ALL. Please help. Problem: video calls drop after 15-20 seconds all the time. Use * latest cvs. from sip.conf [1102] type=friend username=1102 host=dynamic callerid=Veo webcam<1102> canreinvite=no disallow=all allow=gsm ;allow=ulaw allow=h261 allow=h263 from iax.conf [peer2] ; 192.168.0.7 type=friend port=4569 auth=md5 secret=second2 context=local host=dynamic qualify=yes trunk=yes